In recent years, the telecommunications industry has witnessed the proliferation of vocoders in order to meet bandwidth demands of different wireline and wireless communications systems. The name <<vocoder>> stems from the fact that its applications are specific to the encoding and decoding of voice signals primarily. Vocoders are usually integrated in mobile telephones and the base stations of the communications network. They provide speech compression of a digitized voice \signal as well as the reverse transformation. Thus a vocoder includes an encoder stage that will accept as input a digitized voice signal and output a compressed signal. As for the reverse transformation the vocoder is provided with a decoder stage that will accept the compressed speech signal and output a digitized signal, such as PCM samples.
The main advantage of compressing speech is that it uses less of the limited available link bandwidth for transmission. The main disadvantage is loss of speech quality.
The rapid growth in the diversity of networks and the number of users of such networks is increasing the number of instances where two vocoders are placed in tandem to serve a single connection. In such a case, a first encoder is used to compress the speech of the first mobile user. The compressed speech is transmitted to a base station serving the local mobile where it is decompressed (converted to PCM format samples). The resulting PCM samples arrive at the base station serving the second mobile terminal, over the digital trunk of the communication network, where a second encoder is used to compress the input signal for transmission to the second mobile terminal. A speech decoder at the second mobile terminal decompresses the received compressed speech data to synthesize the original speech signal from the first mobile terminal.
It is well known that tandem of speech codecs usually degrades the voice quality and increases the transmission delay over a communications link. In an attempt to eliminate the condition of vocoder tandeming in a communications link between the first mobile terminal and the second mobile terminal, a method called <<bypass>> has been proposed in the past. The bypass mechanism allows a communications link between the first mobile terminal and the second mobile terminal to acquire different settings namely an active setting and a bypass setting. In use, a digital signal processor associated with the first base station that receives the RF signal from a first mobile terminal determines, through signaling and control that an interoperable codec exists at the second base station associated with the mobile terminal at which the call is directed. The digital signal processor associated with the first base station rather than converting the compressed speech signals into PCM samples invokes the bypass mechanism which sets the communications link in the bypass setting and outputs the compressed speech in the network towards the second base station. The compressed speech signal, when arriving at the digital signal processor associated with the second base station is routed such as to bypass the local vocoder. Decompression of the signal occurs only at the second mobile terminal. Tandem free operation (TFO) standard, such as TIA/EIA-829, GSM 08.62 have been developed to allow the removal of intermediate compression/decompression stages in the base stations.
The communications link between the first and second mobile terminals frequently includes functional processing stages providing audio processing operations for improving the audio quality of an uncompressed signal. Examples of such audio processing operations include echo cancellation, level adjustment and noise reduction among others. Such functional processing operations may be located within the base stations or in locations in between the base stations. In general such processing operations can only be applied to a decompressed audio signal and as such cannot be applied when the communications link is in the bypass setting. Depending on the nature and the configuration of the call, the omission of these processing operations can at times reduce the overall quality of service by a larger amount than the voice quality gained due to the removal of tandem codecs. Consequently, a deficiency in the above-described bypass method is that in certain cases the bypass mechanism may result in an overall reduction in the quality of the audio signal.
Consequently, there is a need in the industry for providing a method and apparatus for controlling an operative setting of a communications link that at least in part alleviates the problems associated with the prior art.